Hearing device with suppression of sound impulses

ABSTRACT

A hearing device includes: at least one microphone for converting sound received by the at least one microphone into an audio signal; a sound impulse detector configured for detecting a presence of an impulse in the audio signal; and a signal processor configured for processing the audio signal into a processed audio signal in response to the presence of the impulse in the audio signal as detected by the sound impulse detector; and a receiver coupled to the signal processor for converting the processed audio signal into an output sound signal for emission towards an eardrum of a user; wherein the sound impulse detector is configured for operation in a frequency domain for detecting presence of the impulse in the audio signal.

RELATED APPLICATION DATA

This application is a continuation of U.S. patent application Ser. No.15/053,558, filed on Feb. 25, 2016, pending, which claims priority to,and the benefit of, European Patent Application No. 15202409.7, filed onDec. 23, 2015, pending. The entire disclosure of the above applicationis expressly incorporated by reference herein.

FIELD

A new hearing device is provided capable of suppressing sound impulsesfor ear protection and user comfort.

BACKGROUND

Hearing impaired persons are, compared to persons with normal hearing,more susceptible to discomfort when subjected to sound impulses of highsound pressure levels. Known hearing aids comprise compressors thatutilize dynamic sound level compression with time constants that aresufficiently long to avoid distortion of temporal characteristics ofspeech. The associated recruitment effect combined with a hearing aidincreases the discomfort caused by sound impulses with high energy.

SUMMARY

A new hearing device and method are provided that alleviates discomfortcaused by sound impulses. Sound impulses are sounds exhibiting highsound pressures during a short time period, such as a time period in theorder of milliseconds, such as shorter than 10 milliseconds.

The new method comprises the steps of

converting sound into an audio signal,

subjecting the audio signal to a frequency transformation,

detecting presence of an impulse in the audio signal based on thefrequency transformed audio signal, and

processing the audio signal into a processed audio signal in response todetected presence of the impulse in the audio signal,

converting the processed signal into an output sound signal, and

emitting the output sound signal towards an eardrum of a human.

The frequency transformation may be a warped frequency transformation.

The frequency transformation may be a Warped Fourier Transformation, aWarped Discrete Fourier Transformation, a Warped Fast FourierTransformation, etc.

The warped frequency bands may correspond to the Bark frequency scale ofthe human ear.

The frequency transformation may be a non-warped frequencytransformation.

The frequency transformation may be a Fourier Transformation, such as aDiscrete Fourier Transformation, a Fast Fourier Transformation, etc.

The new hearing device comprises

at least one microphone for converting sound received by the at leastone microphone into an audio signal,

a sound impulse detector configured for detecting presence of an impulsein the audio signal, and

a signal processor configured for processing the audio signal into aprocessed audio signal in response to presence of the impulse in theaudio signal as detected by the sound impulse detector, and

a receiver connected to an output of the signal processor for convertingthe processed signal into an output sound signal for emission towards aneardrum of a user, and

wherein

the sound impulse detector is configured for operation in the frequencydomain, e.g. utilizing a Fourier Transformation, such as the DiscreteFourier Transformation, the Fast Fourier Transformation, etc., fordetecting presence of the impulse in the audio signal.

The sound impulse detector may be configured for utilizing a warpedfrequency transformation, such as the Warped Fourier Transformation, theWarped Discrete Fourier Transformation, the Warped Fast FourierTransformation, etc., for transforming the audio signal into a warpedfrequency domain.

The warped frequency bands may correspond to the Bark frequency scale ofthe human ear.

The sound impulse detector may be configured for determining a signallevel S₀ of the audio signal in a frequency band F_(i) at a time t₀ andcomparing the determined signal level S₀ with a signal level S⁻¹ basedon at least one previously determined signal level in the frequency bandF_(i) when determining presence of the impulse in the audio signal.

The sound impulse detector may be configured for determining presence ofthe impulse in the audio signal when the ratio between the signal levelS₀ of the audio signal in a frequency band F_(i) determined at time toand the signal level S⁻¹ based on at least one previously determinedsignal level in the frequency band F_(i) is greater than a predeterminedthreshold Th_(i) for a predetermined number N of frequency bands F_(i).

The signal level may be the sound pressure level (SPL) in dB, i.e. theratio of the root mean square sound pressure and a reference soundpressure of 20 μPa in dB.

Compared to speech, a sound impulse causing discomfort to a humantypically exceeds the predetermined threshold in a large number offrequency bands, such as in a number of frequency bands larger than halfthe total number of frequency bands, for example 10 for a total numberof 17 frequency bands, i.e. N may be equal to 10 for a total number of17 frequency bands.

The threshold may be equal to 10 dB for all frequency bands.

The hearing device may further comprise a sound environment detector forclassifying the sound environment into a predetermined set of soundenvironment classes.

The sound impulse detector may be configured for operation in responseto the sound environment class determined by the sound environmentdetector, for example the threshold Th_(i) may be a function of thesound environment class determined by the sound environment detector.

A broad-band power level may also be included in the determination ofpresence of an impulse in order to further distinguish presence of animpulse over the on-set of speech. For example, determination ofpresence of an impulse may require that the total sound pressure levelof the frequency transformed audio signal is larger than a predeterminedthreshold, such as 75 dB_(SPL), 80 dB_(SPL), etc.

The predetermined threshold value may be adjusted in accordance withuser preferences, as explained below in connection with table 1 which isreproduced from W. O. Olsen: “Average speech levels and spectra invarious speaking/listening conditions, a summary of the Pearson,Bennett, Fidell (1977) report,” American Journal of Audiology, vol. 7,pp. 21-25, 1998.

The hearing device may further comprise a sound impulse suppressorconfigured for suppressing impulses detected by the sound impulsedetector.

The sound impulse suppressor may be configured for attenuating theimpulse in one or more frequency bands, such as all of the frequencybands of the hearing device.

The sound impulse suppressor may be configured for attenuating theimpulse with an amount that is a function of the sound environment classdetermined by the sound environment detector.

The sound impulse suppressor may be configured for attenuating theimpulse in such a way that the receiver does not emit sound, orsubstantially does not emit sound, originating from the impulse. Forexample, if a user wears a hearing aid with the sound impulse detectorand the sound impulse suppressor, the sound impulse suppressor may beconfigured for attenuating the impulse in such a way that the user hearsthe corresponding sound impulse as if the user did not wear the hearingaid.

Various signal processing parameters, such as detection thresholds,attenuation levels, etc., of the new sound impulse detector and soundimpulse suppressor may be adjustable in accordance with user inputs.

The hearing device may be a hearing aid, such as a BTE, RIE, ITE, ITC,or CIC, etc., hearing aid including a binaural hearing aid.

The hearing device may be a headset, headphone, earphone, ear defender,or earmuff, etc., such as an Ear-Hook, In-Ear, On-Ear, Over-the-Ear,Behind-the-Neck, Helmet, or Headguard, etc.

For example, the new hearing device is a new hearing aid comprising ahearing loss processor that is configured to process the audio signal inaccordance with a predetermined signal processing algorithm to generatea hearing loss compensated audio signal compensating a hearing loss of auser.

Processing, including signal processing, in the new hearing device maybe performed by dedicated hardware or may be performed in a signalprocessor, or performed in a combination of dedicated hardware and oneor more signal processors.

As used herein, the terms “processor”, “central processor”, “messageprocessor”, “signal processor”, “controller”, “system”, etc., areintended to refer to CPU-related entities, either hardware, acombination of hardware and software, software, or software inexecution.

For example, a “processor”, “signal processor”, “controller”, “system”,etc., may be, but is not limited to being, a process running on aprocessor, a processor, an object, an executable file, a thread ofexecution, and/or a program.

By way of illustration, the terms “processor”, “central processor”,“message processor”, “signal processor”, “controller”, “system”, etc.,designate both an application running on a processor and a hardwareprocessor. One or more “processors”, “central processors”, “messageprocessors”, “signal processors”, “controllers”, “systems” and the like,or any combination hereof, may reside within a process and/or thread ofexecution, and one or more “processors”, “central processors”, “messageprocessors”, “signal processors”, “controllers”, “systems”, etc., or anycombination hereof, may be localized in one hardware processor, possiblyin combination with other hardware circuitry, and/or distributed betweentwo or more hardware processors, possibly in combination with otherhardware circuitry.

A hearing device includes: at least one microphone for converting soundreceived by the at least one microphone into an audio signal; a soundimpulse detector configured for detecting a presence of an impulse inthe audio signal; and a signal processor configured for processing theaudio signal into a processed audio signal in response to the presenceof the impulse in the audio signal as detected by the sound impulsedetector; and a receiver coupled to the signal processor for convertingthe processed audio signal into an output sound signal for emissiontowards an eardrum of a user; wherein the sound impulse detector isconfigured for operation in a frequency domain for detecting presence ofthe impulse in the audio signal.

Optionally, the sound impulse detector is configured for utilizing anon-warped frequency transform for transforming the audio signal into anon-warped frequency domain.

Optionally, the sound impulse detector is configured for utilizing alinear frequency transform for transforming the audio signal into alinear frequency domain.

Optionally, the sound impulse detector is configured for determining asignal level S₀ of the audio signal in a frequency band F_(i) at a timeto, and comparing the determined signal level S₀ with a signal levelbased on at least one previously determined signal level in thefrequency band F_(i) when detecting the presence of the impulse in theaudio signal.

Optionally, the sound impulse detector is configured for detecting thepresence of the impulse in the audio signal when a ratio between thesignal level S₀ of the audio signal in the frequency band F_(i)determined at time to and the signal level S⁻¹ that is based on the atleast one previously determined signal level in the frequency band F_(i)is greater than a predetermined threshold Th_(i) for a predeterminednumber N of bands in the frequency Band F_(i).

Optionally, the sound impulse detector is configured for detecting thepresence of the impulse in the audio signal when a ratio between thesignal level S₀ being a sum of the audio signal in a plurality offrequency bands F_(i), F_(i+1) determined at times t_(i), t_(i+1) andthe signal level S⁻¹ being a sum based on a plurality of previouslydetermined signal level in the frequency bands F_(i), F_(i+1) is greaterthan a predetermined threshold Th_(i) for a predetermined number N ofbands in the frequency bands F_(i), F_(i+1).

Optionally, the sound impulse detector is configured for detecting thepresence of the impulse in the audio signal when a broad-band powerlevel of the audio signal is higher than a power threshold level.

Optionally, the hearing device further includes a sound environmentdetector for classifying a sound environment into a sound environmentclass, and wherein the sound impulse detector is configured foroperation in response to the sound environment class determined by thesound environment detector.

Optionally, the hearing device further includes a sound environmentdetector for classifying a sound environment into a sound environmentclass, and wherein the sound impulse detector is configured foroperation in response to the sound environment class determined by thesound environment detector; wherein the threshold Th_(i) is a functionof the sound environment class determined by the sound environmentdetector.

Optionally, a signal processing parameter of the sound impulse detectoris adjustable in accordance with a user input.

Optionally, the hearing device further includes a sound impulsesuppressor configured for attenuating the impulse in response to thepresence of the impulse as detected by the sound impulse detector.

Optionally, the hearing device further includes a sound impulsesuppressor configured for attenuating the impulse in response to thepresence of the impulse as detected by the sound impulse detector;wherein the sound impulse suppressor is configured for attenuating theimpulse with an amount that is a function of the sound environment classdetermined by the sound environment detector.

Optionally, the sound impulse suppressor is configured for attenuatingthe impulse in such a way that the receiver does not emit soundoriginating from the impulse.

Optionally, a signal processing parameter of the sound impulsesuppressor is adjustable in accordance with a user input.

Optionally, the hearing device is a hearing aid, and wherein the signalprocessor comprises a hearing loss processor that is configured toprocess the audio signal in accordance with a predetermined signalprocessing algorithm to generate a hearing loss compensated audio signalcompensating a hearing loss of the user.

Optionally, the hearing loss processor comprises a dynamic rangecompressor.

Optionally, the hearing device is a hearing protector comprising apassive dampener configured for dampening sound, and wherein at least apart of the passive dampener is configured for occluding a part of anear canal of the user.

A method includes: converting sound into an audio signal; subjecting theaudio signal to a frequency transformation to obtain a frequencytransformed audio signal; detecting a presence of an impulse in theaudio signal based on the frequency transformed audio signal; processingthe audio signal into a processed audio signal in response to thedetected presence of the impulse in the audio signal; converting theprocessed signal into an output sound signal; and emitting the outputsound signal towards an eardrum of a human.

Other and further aspects and features will be evident from reading thefollowing detailed description of the embodiments.

BRIEF DESCRIPTION OF THE DRAWINGS

The drawings illustrate the design and utility of embodiments, in whichsimilar elements are referred to by common reference numerals. Thesedrawings are not necessarily drawn to scale. In order to betterappreciate how the above-recited and other advantages and objects areobtained, a more particular description of the embodiments will berendered, which are illustrated in the accompanying drawings. Thesedrawings depict only typical embodiments and are not therefore to beconsidered limiting of its scope.

In the drawings:

FIG. 1 shows a block diagram of a signal processing scheme of a priorart hearing aid,

FIG. 2 shows a plot of delay as a function of frequency in a prior artwarped delay line,

FIG. 3 shows a block diagram of a signal processing scheme according tosome embodiments,

FIG. 4 shows a plot of warped frequency bands,

FIG. 5 shows a plot of frequency bands of an sound impulse detectoraccording to some embodiments,

FIG. 6 shows a plot of gain reduction as a function of broadband poweraccording to some embodiments,

FIG. 7 shows plots of impulse detection and gain reduction as a functionof time according to some embodiments,

FIG. 8 shows a flow-chart of power estimation calculation according tosome embodiments,

FIG. 9 shows plots of impulse detection and α-values as a function oftime according to some embodiments,

FIG. 10 shows a plot of rise power thresholds for different soundenvironments according to some embodiments, and

FIG. 11 shows a block diagram of a signal processing scheme according tosome embodiments.

DETAILED DESCRIPTION

Various illustrative examples of the new hearing device according to theappended claims will now be described more fully hereinafter withreference to the accompanying drawings, in which various embodiments ofnew hearing device are illustrated. The new hearing device according tothe appended claims may, however, be embodied in different forms andshould not be construed as limited to the embodiments set forth herein.In addition, an illustrated embodiment needs not have all the aspects oradvantages shown. An aspect or an advantage described in conjunctionwith a particular embodiment is not necessarily limited to thatembodiment and can be practiced in any other examples even if not soillustrated, or if not so explicitly described.

As used herein, the singular forms “a,” “an,” and “the” refer to one ormore than one, unless the context clearly dictates otherwise.

FIG. 1 schematically illustrates a prior art hearing aid signalprocessing scheme 10 with dynamic signal compression performed in ahearing aid compressor well-known in the art of hearing aids.

The known hearing aid compressor performs a warped frequencytransformation and controls the gain in 17 warped frequency bandscorresponding to the Bark frequency scale of human hearing. The gainsare controlled in accordance with the fitting rule of the hearing aidand the hearing loss of the user of the hearing aid so that the dynamicrange of a human with normal hearing is compressed into the residualdynamic range of the user with a hearing loss resulting in loss ofdynamic range as is well-known in the art of hearing aids. The attackand release time constants are quite long in order to avoid distortionof speech.

The trade-off is that short, intense sounds might be over-amplified andin combination with the rapid increase in perceived loudness, also knownas recruitment, this could potentially be a downside of the hearing aidcompressor.

Due to the nature of sound impulses, such as door slamming, clinking ofsilverware, jangling of keys, etc., hearing aid users are often leftwith discomfort and annoyance in their daily usage.

In many cases a very rare occurring event, influences the hearing deviceusage in such a way, that the hearing impaired user might lose all theintended benefits from wearing the devices. Turning down the volume orslightly removing the hearing device from the ear, which to some extendis similar to a frequency dependent gain reduction, is something that analgorithm should be able to do both faster and more effective. In orderto obtain suitable impulse suppression, impulse detection and responsehave to be performed with minimum delay, e.g. maintaining un-assistedloudness during the impulse.

For mild hearing losses, protecting against sound impulses could alsohave another effect; preserving hearing. Persons with normal hearinghave what is sometimes referred to as the acoustic reflex which isinitiated by high sound pressure levels (SPL). It selectively reducesthe intensity of sound transmitted to the inner ear; however with ashort delay of approximately 20 ms. Hence, high level impulse soundssuch as gun shots may be too short for the muscle to react to, resultingin possibly permanent hearing loss.

Hearing device users with certain combinations of hearing loss andconfigurations are also disturbed more by less intensive soft soundimpulses. This could be the clicking of a computer's keyboard, orrustling paper.

The new sound impulse detector and/or sound impulse suppressor may beadjustable in accordance with user inputs.

In the illustrated embodiment, gain adjustments are performed taking thecurrent gain settings of the hearing aid compressor into account.

The known warped hearing aid compressor signal processing scheme isillustrated in a high level in FIG. 1. The numbering indicates the orderof execution within one block of samples. The delay from input to outputof the compressor is equal to the time of sampling one block of samples,e.g. a few milliseconds.

Estimating power with critical band resolution is achieved by warpingthe delay line. The all-pass filters serve to implementfrequency-dependent unit delays, low frequencies are stretched and highfrequencies are compressed. The group-delay as illustrated in FIG. 2,for a high bandwidth configuration, is low at high frequencies while thelow frequency area is exposed to a longer group delay. It can beobserved that for a compressor system based on the state remaining fromthe last input sample in each block, the group-delay at high frequenciesis much lower than the block rate˜1.5 mS.

In other words, there is a risk that a sound impulse detector based onthe warped delay-line potentially underestimates the high frequency partof blocks 2 with an impulse. High bandwidth platforms have a slightlydifferent MPO implementation compared to the normal bandwidth platforms.The MPO has been updated to avoid sudden changes in the static gainoperation. A high bandwidth MPO partially applies the static gainchanges in intervals of two samples; the full gain change is appliedwithin one block of samples. An impulse gain reduction build on top ofthe existing MPO, would further imply a change in order to deal with thegain update-delay in the direct sound path.

In order to be able to attenuate impulses, a sound impulse detector isadded to the dynamic hearing aid compressor.

The signal processing scheme of a combined sound impulse detector, gainadjustment, and dynamic hearing aid compressor is shown in FIG. 3.

Comparing the execution order of the submodules indicates that the Gaincontroller and Filter Design are now executed before the direct pathprocessing. The Warp Power is still based on the previous blocks and allgain agents are still processing the same data as before i.e. thedynamic hearing aid compressor is not changed. A new Gain Calculationblock has been added before the Gain controller, and an instant changeof frequency response can be obtained. If the sound impulse detector andthe Gain Calculation block are disabled, the illustrated processingscheme will be identical to the processing scheme shown in FIG. 1, i.e.the processing scheme of the known dynamic hearing aid compressor.

Detecting sound impulses in the frequency domain is performed utilizinga second frequency domain transformation. Addressing complexity,resolution and flexibility, the linear DFT in equation (1) is thestarting point for the sound impulse detector.

$\begin{matrix}{{X\lbrack n\rbrack} = {\sum\limits_{k = 0}^{N - 1}{{x(k)}e^{{- j}\frac{2\pi}{N}{nk}}}}} & (1)\end{matrix}$

Preferably, the sound impulse detector should work on the unprocessedinput block. This is illustrated in FIG. 3, where (2) indicatesfrequency domain transformation, following the new arrived input blockof audio samples. Particularly it is of interest how the power risesover time, when looking for impulse patterns. Equation (2) shows thefrequency domain power estimate P[n] of the current block.

P[n]=abs(X[n])²  (2)

Input blocks of samples, that exhibit impulsive nature must have anapproximately instant rise time. In addition, the impulsivecharacteristic causes a power distribution that spans many bands. Asmoothed version of the power estimates per bands˜P [n] is used for theinstant rise feature extraction. The parameter α in equation (3) shouldbe chosen sufficiently small, in order to explore the instant rise timeof the impulse relative to a short history of background power.

For an optimized performance during repetitive impulses, the smoothedpower estimates is not allowed to be updated during detected impulses.In addition, the ability to efficiently track the impulse relies on thepossibility to compare the frequency domain power of the impulse withthe energy just before the impulse onset. Dividing the current powerestimate with the smoothed version as in equation (4), can be used as ameasure of how much the power in the different bands has raised with thenew block of samples.

$\begin{matrix}{{r(n)} = \frac{P\lbrack n\rbrack}{\overset{\_}{P}\lbrack n\rbrack}} & (4)\end{matrix}$

For implementation complexity reasons, the rise measure r (n), couldadvantageously be implemented in the log₂ domain. The precision of thelog₂ is found to be accurate enough, and the remaining part of the soundimpulse detector could improve by having decision and thresholdimplemented in the logarithmic domain, equation (5).

r(n)=log₂(P[n])−log₂({grave over (P)}[n])  (5)

It could be argued that, due to the window size, the power estimates arepoor for the lowest bands. For simplicity and in order to align with theexisting hearing device platform the number of bands L is defined asequation (6)

$\begin{matrix}{L = {\frac{N}{2} + 1}} & (6)\end{matrix}$

where N is the size of the DFT and accordingly, in a non-overlapimplementation, is equal to the processing block-size. Now, a vectorr_(t) build of L bands rise measures in the log₂ domain can beconstructed

r _(t) =[r(0),r(1), . . . ,r(L−1)]  (7)

wherein t is block rate which for a high bandwidth hearing deviceplatform is

$\begin{matrix}{T_{block} = {{N \cdot \frac{1}{fs}} \approx {1.5\mspace{14mu} {ms}}}} & (8)\end{matrix}$

In effect the block rate in eq. (8) also sets the lower limit of theimpulse rise time that the sound impulse detector can observe. Keepingin mind that this limit is not to be confused with the scheme in FIG. 3,which can apply gain reduction instantly with no delay from thedetection point. One major point of concern for a sound impulse detectorwill always be whether it distinguishes between impulse sounds like doorslamming, cutlery etc. and speech onset, which is the portion ofvocalization where impulse-like characteristics can be ascertained. Oneway of addressing this issue could be to include a threshold that wouldoperate on the vector r_(t). Now eq. (9) defines a measure of how manybands in the present power estimate exceeds this threshold.

R _(t)=sum(r _(t)>RiseThreshold)  (9)

The threshold in eq. (9) would be defined in the log₂ domain. Comparedto speech, impulse noises that are annoying in nature, for hearingdevice users, tends to span power over most of the frequency powerbands. Defining that the sum of power bands with instant rise time R_(t)should be above 10, adds another dimension to the task of addressingspeech vs. impulse noise in the sound impulse detector. At this point atrue/false parameter of impulse detection is available.

A final broadband power threshold is also applied to ensure that onlyimpulsive blocks above a particular sound-pressure level are detected.This threshold is applied in order to configure the sensitivity of thesound impulse detector. For end-users that only find intense impulseslike door slams annoying, this threshold can be increased compared tousers who are disturbed by more weak impulses, defined like the clickingof a computer keyboard, clattering dishes etc. For example firecrackerscan reach level as high as 180 dB_(SPL).

Table 1 below shows the speech levels (non-weighted SPL) of casual,normal, raised, loud, and shouted speech by males, females, andchildren:

TABLE 1 Casual Normal Raised Loud Shouted Females 54 58 65 72 82 Males56 61 68 77 89 Children 56 61 67 75 82

A broadband power threshold of the sound impulse detector has anaturally lower limit as indicated in table 1 which is reproduced fromW. O. Olsen: “Average speech levels and spectra in variousspeaking/listening conditions, a summary of the pearson, bennett, fidell(1977) report,” American Journal of Audiology, vol. 7, pp. 21-25, 1998.

In order to apply even more robustness towards knowing the differencebetween speech onset and targeted impulse sounds, this threshold must beset high enough to operate on top of the normal speech production area.The pseudo code in Algorithm 1 summarizes impulse detection of the soundimpulse detector. The output parameter of the detection algorithm isdetect, which holds values between zero and one (0 detect 1). For detectto reach zero after an impulse has decayed to a state where it is nolonger exploring impulsive characteristics, or does not longer complywith the broadband power threshold, a logarithmic release time isapplied. The parameter α_(detect) is used to specify the release time,while the attack time of detect is instant.

A frequency-warped FIR filter can be designed by replacing the unitdelays in the conventional FIR filter with all-pass filter sections. Itserves to match the frequency resolution of the compression system tothe resolution of the human auditory system. Additionally the warpedfilter has a higher group-delay at low frequencies than a conventionalfilter for the same low frequency resolution. As discussed earlier, theshort delay at high frequencies is problematic for a sound impulsedetector e.g. under-sampling can lead to false detection. In addition,the frequency resolution of a DFT based on a warped delay line can limitthe performance of the detection scheme as well. The warp Compressorsystem, or more important the power estimator, is based on the warpeddelay line utilizing the all-pass transfer function in equation 10.

$\begin{matrix}{{H(z)} = \frac{a + z^{- 1}}{1 + {az}^{- 1}}} & (10)\end{matrix}$

where a is the warping parameter. Combined with the warp window thisleads to the 17 bands illustrated in FIG. 4. The warped frequency scalegives a much better match to auditory perception compared to a linearbased system. However, serving to detect and differentiate impulsenoises from the daily sound environment including own and surroundingspeech, the warp-based DFT delivers poor performance. In order to usethe number of frequency bins with instant power rise as a feature fordetecting impulsive input blocks, a much better resolution is needed inthe highest bins. In addition the warp window is constructed to smearadjacent bins to avoid drastic gain differences by the filter designer.The sound impulse detector utilizes a 32-point linear FFT with a Tukeywindow. FIG. 5 illustrates the frequency resolution of the 17 bands.This configuration will not favour speech-like signals. Another choicecould be to use a warped delay line with a positive warping factor. Thiswould further increase the resolution of the highest bins, leading to adetection even more focused on instant power increase in regions notdominated by speech. The primary disadvantage of a detection schemebased on a parallel warped delay line is the computational cost ofreplacing unit delays with first-order all-pass filters.

3.2 Spectral Leakage

The DFT implicit assumes that the signal is periodic in the time frame.When the input block is not periodic then leakage occurs. Leakageresults in misleading information about the spectral amplitude andfrequency. For the sound impulse detector, the worse impact is leakageto adjacent bins, which might lead to false detection. The sound impulsedetector relies on identification of bands with rapid increase of soundpower; spectral leakage contributes to the risk of false detection. ADFT window can be applied to reduce the effects of leakage.

{circumflex over (x)}(n)=x(n)w(n)  (11)

{circumflex over (x)} _(t) =[{circumflex over (x)}(0),{circumflex over(x)}(1), . . . ,{circumflex over (x)}(N−1)]  (12)

X _(t) =DFT({circumflex over (x)} _(t))  (13)

The Gain calculation block may reduce broad-band gain, e.g. the gain inall of the frequency bands, in a plurality of the frequency bands, suchas in more than half of the frequency bands, of the compressor in orderto attenuate the impulse.

The Gain calculation block may restore natural loudness of signals likeslamming doors, clinking of silverware or jangling of keys, in responseto impulse detection. These are all examples of sounds that are part ofthe daily sound environment, but in most cases will generate anunnatural and painful representation at the ear-drum of the hearingdevice user. Focusing on the end-user and what causes the discomfort,the Gain Calculation block must be able to address theover-amplification of short duration impulsive signals. Most likely theun-natural reproduced segments is causes by the linear part of theprescribed gain i.e. the G₅₀ gain is applied for high energy impulsesignals. In other words, what causes the discomfort is end-userdependent and most likely described by the G₅₀ gains. This also meansthat the sound impulse suppressor needs to control gain independently inthe 17 frequency bands, in order to match the behaviour of the warpsystem.

The sound impulse suppressor is configured for attenuating the impulseto a comfort level still being descriptive of the acoustic environment.A very simple approach that does not add significant complexity to therun-time part of the algorithm could be to utilize a gain look-up table.A look-up table would map the broadband power of an impulse section, toa reduction vector, with the needed gains for the 17 warped bands. Agiven fitting rule is used to reach the prescribed gain based on thehearing threshold. In a two power bands configuration, the prescribedgain is implemented by the target G₅₀ and G₈₀ gains. Define a broadbandpower threshold vector B as a starting point

B=[b(0),b(1), . . . ,b(P−1)]  (14)

where P is the power table size i.e. the resolution of the steps thatcan be achieved. The span of power, or the SPL area that sound impulsesuppressor should work within is defined as

power_span=B[P−1]−B[0]  (15)

The target gains are now mapped linearly into this area by means of theparameters min reduction and max reduction. Where min reduction in dBdefines the reduction at the lower boundary of the B vector and maxreduction defines the reduction at the top of the vector. E.g. it isdefined how much of the target gains, G₅₀, that the sound impulsesuppressor will correct for at a given SPL. Use the relativedistribution of broadband power level thresholds AB in order tonormalize this vector

$\begin{matrix}{\hat{B} = \left\lbrack {0{{cumsum}\left( \frac{\Delta \; B}{power\_ span} \right)}} \right\rbrack} & (16)\end{matrix}$

The normalized vector {circumflex over (B)} can be used to linearlyinterpolate from the two-dimensional space defined by min reduction andmax reduction, into the dimension of the B vector. The outcome is avector with gain reduction ratios, in dB, per broadband power level.These reduction numbers are relative to the G₅₀ target gains and thefinal the sound impulse suppressor gains are now defined as a P by 17matrix G. If min reduction is set to 6 dB, the sound impulse suppressorwill apply half of the target gain in reduction during an impulse withthe lowest broadband power. This will then linearly increase up to e.g.max reduction set to 0 dB, where the sound impulse suppressor willreduce the gains equal to the target gains i.e. fully compensate for theAGCI (Automatic Gain Control-Input) gains. FIG. 6 illustrates how thetarget G₅₀ gains are mapped to the sound impulse suppressor gainreductions. This example has the broadband power threshold vector B setto

B=[86 90 94 96 100 100][dBSPL]  (17)

and the target G₅₀ gains used was

G ₅₀=[7 7 7 7 7 7 7 9 10 11 12 14 16 18 26 33 34][dB]  (18)

With min reduction set to 6 dB and max reduction set to 0 dB, it isobserved how the gain reduction gradually increases from half the G₅₀target gains, at an impulse broadband power of 86 dB SPL, up to fullcompensation at 110 dB SPL. When maximum broadband power is reached inthe B vector, the sound impulse suppressor gain reduction is locked tothis level. In addition, the broadband power threshold used in thedetection part should be the same value as the first entry of the Bvector. This will align the sound impulse detector and the gaincalculation block with respect to active area of operation.

In the attempt of securing listening comfort for a broad representationof hearing threshold fittings, the ability of adjusting the sensitivityof the sound impulse detector is needed. Users might express specialneeds and annoyance levels e.g. some hearing impaired might feeldiscomfort even for less intensive impulse-like sounds like clicking ofa computers keyboard, rustling paper etc. There might also be a need fordifferent sensitivity in order to address acclimatization for first-timehearing device users. A simple mild, medium and strong approach ispreferred. This can be achieved by addressing the broadband power levelsduring impulses different, i.e. by defining the vector B per mode. Anexample of how the sound impulse detector modes could be configured isshown in table 2 listing sound impulse detector modes (mild, medium,strong) aligned with broadband power thresholds dB SPL.

TABLE 2 low . . . . high Mild 90 94 98 100 104 114 Medium 86 90 94 96100 110 Strong 75 78 80 84 86 90

In combination with the B vector being set per mode, max reduction andmin reduction could also be included. This enables the sound impulsedetector and sound impulse suppressor to define modes by means of thelevels of where to reduce gains, and indeed also how much to reducegain.

When dealing with discomfort, by reducing gain during impulse sounds,the sound impulse suppressor applies the smallest attack timeachievable. This is possible as already observed in the re-arranged warpsystem in FIG. 3. The broadband power is expected to vary during animpulse; the impact could be that the gain reduction applied willfluctuate causing distortion. This potential issue increases with moreextreme settings of the modes in table 2, e.g. if a mode spans a largearea of sound pressure levels. A way of addressing fluctuating soundimpulse suppressor gains could be to apply an impulse onset detectionparameter. In FIG. 7 (A) this is illustrated. An impulse onset detect isdefined as being the point in time where the previous block was notdetected as part of an impulse sequence, and an impulse is detected inthe present block.

This is described as

$\begin{matrix}{{onset} = \left\{ \begin{matrix}{{true},} & {{if}\left( {{{predetect}==0}\&\&{{detect}==1}} \right)} \\{{false},} & {otherwise}\end{matrix} \right.} & (19)\end{matrix}$

Now, the algorithm can distinguish between impulse onset and the part ofthe impulse where all other conditions are still valid i.e. in themiddle part of the impulse. The strategy for how to apply gain reductionis to use symmetric smoothing of the gain in blocks preceding the blockwhere impulse onset is detected. The onset block will determine the gainstarting point according to the current broadband power.

Short impulse-like signals are in some situations part of the spatialawareness experienced by the hearing impaired. In the sense that roomreverberation is providing perceptual awareness about thecharacteristics and size of the room. Optimally, the gain reductionrelease time must be set according to the acoustic environment e.g. withrespect to the reverberation time of the room, hall etc. The releasetime, in combination with the normal AGCI attack time, should be set sothat the early reflections are still suppressed, while late reflectionsare perceived with normal loudness. For speech intelligibility, earlyreflections are very important for both normal hearing and hearingimpaired persons, while the late reflections often degrades the abilityto understand speech in noise. For impulse signals this is opposite, inthe sense that late reflections adds to the perception of the roomcharacteristics. For a hearing device user, early reflections, whichcould still include high energy at some frequencies, would still beover-amplified and though add to the discomfort (given that the AGCIrelease time is long compared to the arrival of the early reflections).

The sound impulse suppressor may have a broadband gain release time,i.e. all bands are configured to the same time constants and thisparameter is not adapted in any way during run-time. During the releasetime the gain reduction provided by the sound impulse suppressor willdecade. This serves to smooth the transition between the sound impulsesuppressor actively reducing the impact of the impulse, and restoringnormal AGCI control of input related gain handling. The release of gainreduction will be based on a threshold on the detect parameter, FIG. 7(A). This parameter can be used in the decision of when the impulse hasdecreased its strength to a point where it can be defined as completed.At this point the gain release takes over, FIGS. 7 (A) and (B)illustrates the usage of a detection threshold.

A way of detecting and reacting upon impulsive inputs has been describedin the previous sections. It is clear that input signals with impulsiveonset and a certain length will have the ability to lock the detectingstate of the algorithm. A measure of the duration of an impulse and amaximum impulse duration definition is needed. In order to hand-oversignals that in nature exploits impulse start conditions, but are muchlonger in duration, the sound impulse suppressor is configured to fadeout and leave the gain handling to the normal warp compressor system. Ifa signal has impulsive onset followed by a long sequence with energy inmany bands, the power estimation will, by design, be locked by the soundimpulse detector. The consequence is that these types of sounds will beattenuated by the Gain Calculation block for much longer time thatrequired, i.e. it will overlap with the normal warp compressor systemwhich over time will reduce gains. E.g. the start of a lawnmower willtypically go from a very quiet condition, over a short impulsive partand then stay noisy in many bands for a longer period. A definition ofthe maximum duration of the impulses the sound impulse suppressor shouldhandle, and how to measure and fade-out is needed. A very elegant way ofcontrolling the sound impulse detector part in relation to the durationof the impulse is to adaptively control the parameter in equation (3).Based on the information of where the current detection estimate is intime, it is possible to control the update rate of the frequency bandpower estimate smoothing. The flow-chart in FIG. 8 illustrates how tocontrol and update the power estimator part of the detector. Based on adefined maximum duration count it is possible to decide the smoothingrate based on the parameter α. An α-value going towards zero will simplystop the smoothing of the frequency bands power estimates. This is thepreferred setting in the sequence following the onset of an impulse,i.e. stop updating. For normal operation, where no impulse is detected arather high value of a is needed in order to base the detection decisionon the history of energy per bands. A fast power update is needed whenthe maximum duration of an impulse is reached. The consequence oflowering the α-parameter, a fast update speed, will be that the powerestimates will quickly adapt to the levels which is currentlyexperienced by e.g. a lawnmower. The difference between the currentestimate and the smoothed estimates will no longer exploit instant riseand the detection scheme will resign to release mode, and we can applynormal a values for a rather slow update rate again. The sequence ofchanging the α-parameter based on the detection value is shown in FIG.9.

At this point the differentiation in attenuation applied by the soundimpulse suppressor is based purely on the broadband power. Gain vectorsbased on the prescribed gain are calculated on-line and appliedaccording to the estimated broadband power. This scheme seems to favourthe situations close to the G₅₀ knee-point, is could be an advantage toinclude another knee-point to reach a stage where the applied gain issteered towards the present sound pressure level. One solution could beto utilize the classifier classes which to some extend includesinformation about the sound pressure level of the environments. Table 3lists the sound pressure levels related to each of the classifier outputclasses. According to the table, it makes sense to add another gaintable and base the calculated gain tables on a knee-point atapproximately 75 dB_(SPL). The classifier environments can now be usedto steer the gain reduction tables in order to achieve that the soundimpulse suppressor takes into account the current estimated soundenvironments. E.g. silent environments, where the prescribed gain are inthe linear area, maps to higher gain reductions and high noiseenvironments, where the gain operates in the compression area, shouldattend less gain reduction from the sound impulse suppressor.

According to another embodiment with a signal processing scheme shown inFIG. 11, e.g. for a hearing protection device, wherein the warped delayline and warped power estimates are not present, a more simple soundimpulse detector and sound impulse suppressor can be utilized. Inaddition applications where the gain reduction is not to be associatedwith a hearing loss or prescribed gain, the impulse detection block ofthe sound impulse detector could provide input to a gain control unitrather than a gain calculation unit of the sound impulse suppressor. Again control unit could control several parameters of the Gaincontroller given inputs from other gain agents and the Impulse Detectionblock.

Although particular embodiments have been shown and described, it willbe understood that they are not intended to limit the claimedinventions, and it will be obvious to those skilled in the art thatvarious changes and modifications may be made without departing from thespirit and scope of the claimed inventions. The specification anddrawings are, accordingly, to be regarded in an illustrative rather thanrestrictive sense. The claimed inventions are intended to coveralternatives, modifications, and equivalents.

1. A hearing device comprising: at least one microphone for convertingsound received by the at least one microphone into an audio signal; asound impulse detector configured for detecting a presence of an impulsein the audio signal; and a signal processor configured for processingthe audio signal into a processed audio signal in response to thepresence of the impulse in the audio signal as detected by the soundimpulse detector; and a receiver coupled to the signal processor forconverting the processed audio signal into an output sound signal foremission towards an eardrum of a user; wherein the sound impulsedetector is configured for operation in a frequency domain for detectingpresence of the impulse in the audio signal.
 2. The hearing deviceaccording to claim 1, wherein the sound impulse detector is configuredfor utilizing a non-warped frequency transform for transforming theaudio signal into a non-warped frequency domain.
 3. The hearing deviceaccording to claim 2, wherein the sound impulse detector is configuredfor utilizing a linear frequency transform for transforming the audiosignal into a linear frequency domain.
 4. The hearing device accordingto claim 1, wherein the sound impulse detector is configured fordetermining a signal level S₀ of the audio signal in a frequency bandF_(i) at a time t₀, and comparing the determined signal level S₀ with asignal level S⁻¹ based on at least one previously determined signallevel in the frequency band F_(i) when detecting the presence of theimpulse in the audio signal.
 5. The hearing device according to claim 4,wherein the sound impulse detector is configured for detecting thepresence of the impulse in the audio signal when a ratio between thesignal level S₀ of the audio signal in the frequency band F_(i)determined at time t₀ and the signal level S⁻¹ that is based on the atleast one previously determined signal level in the frequency band F_(i)is greater than a predetermined threshold Th_(i) for a predeterminednumber N of bands in the frequency Band F_(i).
 6. The hearing deviceaccording to claim 4, wherein the sound impulse detector is configuredfor detecting the presence of the impulse in the audio signal when aratio between the signal level S₀ being a sum of the audio signal in aplurality of frequency bands F_(i), F_(i+1) determined at times t_(i),t_(i+1) and the signal level S⁻¹ being a sum based on a plurality ofpreviously determined signal level in the frequency bands F_(i), F_(i+1)is greater than a predetermined threshold Th_(i) for a predeterminednumber N of bands in the frequency bands F_(i), F_(i+1).
 7. The hearingdevice according to claim 1, wherein the sound impulse detector isconfigured for detecting the presence of the impulse in the audio signalwhen a broad-band power level of the audio signal is higher than a powerthreshold level.
 8. The hearing device according to claim 1, furthercomprising a sound environment detector for classifying a soundenvironment into a sound environment class, and wherein the soundimpulse detector is configured for operation in response to the soundenvironment class determined by the sound environment detector.
 9. Thehearing device according to claim 5 or 6, further comprising a soundenvironment detector for classifying a sound environment into a soundenvironment class, and wherein the sound impulse detector is configuredfor operation in response to the sound environment class determined bythe sound environment detector; wherein the threshold Th_(i) is afunction of the sound environment class determined by the soundenvironment detector.
 10. The hearing device according to claim 1,wherein a signal processing parameter of the sound impulse detector isadjustable in accordance with a user input.
 11. The hearing deviceaccording to claim 1, further comprising a sound impulse suppressorconfigured for attenuating the impulse in response to the presence ofthe impulse as detected by the sound impulse detector.
 12. The hearingdevice according to claim 8, further comprising a sound impulsesuppressor configured for attenuating the impulse in response to thepresence of the impulse as detected by the sound impulse detector;wherein the sound impulse suppressor is configured for attenuating theimpulse with an amount that is a function of the sound environment classdetermined by the sound environment detector.
 13. The hearing deviceaccording to claim 11, wherein the sound impulse suppressor isconfigured for attenuating the impulse in such a way that the receiverdoes not emit sound originating from the impulse.
 14. The hearing deviceaccording to claim 11, wherein a signal processing parameter of thesound impulse suppressor is adjustable in accordance with a user input.15. The hearing device according to claim 1, wherein the hearing deviceis a hearing aid, and wherein the signal processor comprises a hearingloss processor that is configured to process the audio signal inaccordance with a predetermined signal processing algorithm to generatea hearing loss compensated audio signal compensating a hearing loss ofthe user.
 16. The hearing device according to claim 15, wherein thehearing loss processor comprises a dynamic range compressor.
 17. Thehearing device according to claim 1, wherein the hearing device is ahearing protector comprising a passive dampener configured for dampeningsound, and wherein at least a part of the passive dampener is configuredfor occluding a part of an ear canal of the user.
 18. A methodcomprising: converting sound into an audio signal; subjecting the audiosignal to a frequency transformation to obtain a frequency transformedaudio signal; detecting a presence of an impulse in the audio signalbased on the frequency transformed audio signal; processing the audiosignal into a processed audio signal in response to the detectedpresence of the impulse in the audio signal; converting the processedsignal into an output sound signal; and emitting the output sound signaltowards an eardrum of a human.